Organizations worldwide seek to reduce the rising costs associated with various forms of communications. Efforts to consolidate separate voice, fax and data resources offer an opportunity for significant savings. These organizations are pursuing solutions that will enable them to take advantage of excess capacity on broadband data networks to accommodate voice, fax and data transmissions as an alternative to costlier mediums.
Voice over Internet protocol (VoIP) is an Internet protocol (IP) telephony that refers to voice communication services that are transported via an IP-based data network, such as the Internet, rather than the public switched telephone network (PSTN). IP networks use packet or cell switching technologies in contrast to circuit switching technologies used by the PSTN. Basic steps involved in a VoIP telephone call include conversion of the originating analog signal into a signal having a digital format. Then compression and translation of this digital signal into IP packets allows transmission over the IP network. The process is reversed at the receiving end of the transmission thereby again providing an analog signal for reception.
Session initiation protocol (SIP) is a signaling protocol used for creating, modifying and terminating sessions, such as IP voice calls or multimedia conferences, that have one or more participants in an IP network. SIP is a request-response protocol used in VoIP that closely resembles HTTP and SMTP, which are the two Internet protocols that power the World Wide Web and e-mail, respectively. The SIP user agent and the SIP proxy server are basic components that support the use of SIP. The SIP user agent is effectively the end system component for the call, and the SIP proxy server handles the signaling associated with multiple calls. This architecture allows peer-to-peer calls to be accomplished using client-server protocol.
A media gateway links the packet-switched IP network with the circuit-switched PSTN. The media gateway terminates voice calls on the inter-switched trunks from the PSTN, compresses and forms packets of the voice data and delivers the compressed voice packets to the IP network. For call origination in the IP network, the media gateway performs the reverse of this order. The media gateway controller accomplishes the registration and management of resources (provisioning) at the media gateway.
Measurements associated with SIP proxy servers show that they spend about 25 percent of their total message processing time in parsing. SIP parsers share common data structures and routines that include message data structures, string manipulation and memory allocation with the remaining SIP proxy server code. Currently, SIP parsers process the entire SIP message or examine the message sequentially until the mandatory headers have been analyzed thereby making current SIP parsers CPU-intensive algorithms. State of the art SIP parsers use the same number of CPU cycles per message for the SIP message, independent of their current load. Therefore, current CPU consumption is directly proportional to the SIP message load. Further, CPU and network consumption is directly proportional to the size of the SIP message.
Accordingly, what is needed in the art is an enhanced way to parse SIP messages that is more sensitive to parameters such as message, CPU, disk, memory and network element load conditions.